System for transmitting high quality speech signals on a voice over internet protocol network

ABSTRACT

The VoIP quality speech process is activated when a subscriber accesses a speech quality sensitive resource or in response to an activation of the feature by the subscriber, or when it is determined that the originating subscriber terminal device requires the transmission of high quality speech signals. A transmit buffer, associated with the port circuit that serves the originating device, stores a predetermined number of packets as they are transmitted from the originating device. In the case of lost or damaged packets, the VoIP quality speech system activates the transmit buffer to retransmit the missing or damaged packet to the destination device. Intelligent buffer management is provided, where the destination device can regulate the size of the transmit buffer as well as the size of its jitter buffer.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a divisional of U.S. patent application Ser. No.10/777,933 filed Feb. 12, 2004.

FIELD OF THE INVENTION

This invention relates to an Internet Protocol Network and a system thattransmits high quality speech signals over this network for selectedapplications.

BACKGROUND OF THE INVENTION

It is a problem in the field of Internet Protocol Networks that some ofthe data packets may fail to arrive at their intended destination.Transmission protocols such as TCP/IP permit receiving devices torequest that missing packets be retransmitted; unfortunately, thisretransmission process often results in long pauses in the data stream,as well as data transmission latencies of more than several hundredmilliseconds, thereby rendering schemes such as TCP/IP inappropriate formost telephony applications.

For these reasons, Voice over Internet Protocol (VoIP) systems commonlyuse a transmission scheme called User Datagram Protocol, or UDP. Thismechanism does not suffer from the pauses or transmission latencies thatwould be seen if TCP/IP were used for VoIP, chiefly because, unlikeTCP/IP, there is no retransmission of missing packets. Instead, IPnetworks often try to reduce VoIP packet loss by assigning a higherpriority (commonly referred to as Quality of Service or QoS) to UDPpackets. Concurrently, many VoIP telephones incorporate packet lossconcealment algorithms that try to trick the human ear by replacing themissing packet with data that is extrapolated from the data received orwith data that is commonly referred to as “comfort noise.”

Unless the level of packet loss becomes extreme (on the order of 5% orgreater, depending on the audio encoding algorithm being used), the useof high quality packet loss concealment algorithms allows UDP to be anacceptable transmission protocol for person-to-person voiceconversations. This is because it is relatively easy to trick the humanear into hearing something that isn't there. Unfortunately, the packetloss concealment algorithms of the present art do not mitigate thedeleterious effects of packet loss on many accuracy-sensitiveapplications for which voice channels (and therefore UDP) are commonlyused; examples include automatic speech recognition systems, automaticspeaker identification systems, and the TTY/TDD communication commonlyemployed by people with hearing deficits.

It is of interest to note that applications such as these, which tend tobe very sensitive to the effects of packet loss, tend not to beespecially sensitive to the effects of latency. Illustratively,point-to-point transmission delays on the order of half a second wouldbe unacceptable in a voice conversation between two people, but wouldprobably not be noticeable in a TTY/TDD conversation, or when anindividual is speaking commands to a typical automatic speechrecognition system. In other words, these are applications for which itwould make sense to accept a greater degree of latency in exchange forreduced packet loss.

A superficial analysis of this problem might cause one to conclude thatthe use of TCP/IP for these applications, rather than UDP, might be areasonable solution. Although the use of TCP/IP would provide for theretransmission of missing packets, there are other considerations thatrender this approach impractical. Reasons include:

-   (1) Transitioning back and forth between TCP/IP and UDP on the same    call would be difficult to support from an engineering standpoint,    and is not even permitted within existing Internet standards. An    example of where this might be needed would be a call in which one    of the parties is hearing-impaired, but not deaf; these individuals    often prefer to intermix voice and TTY/TDD on the same call.-   (2) The adding of a resource that requires TCP/IP to a pre-existing    UDP connection would be difficult to support from an engineering    standpoint, and is not even permitted within existing Internet    standards. An example of this type of situation would be a telephone    conversation between two people, in which an automatic speech    recognition resource is added to the call.-   (3) There is no mechanism within TCP/IP to ensure that the    transmission pauses, while waiting for retransmitted packets to    arrive, occur in places where they will do no harm to an audio    stream (e.g., between spoken words, rather than within a word, or    between TTY characters, rather than within a character).-   (4) If audio packets are tagged as TCP/IP, rather than UDP, VoIP QoS    mechanisms within the Internet may fail to classify these as high    priority packets, thereby exacerbating the packet loss problem even    further.    These and other problems are addressed by the disclosures contained    herein.

BRIEF SUMMARY OF THE INVENTION

The above-described problems are solved and a technical advance achievedby the present system for transmitting high quality speech signals on aVoice over Internet Protocol Network, termed VoIP quality speech system.This VoIP quality speech system selectively activates a speechtransmission mode that tolerates transmission delays of several hundredmilliseconds, since these delays are not noticeable in manyapplications, especially those where the speech transmissions are oneway in nature. For applications where speech quality is of paramountimportance, then speed is sacrificed in this mode, and the associatedInternet Protocol Network is activated to automatically adjust itsencoding and transmission characteristics for these applications. Forexample, voice and TTY speech signals are encoded in a high qualityformat and at no point would the Internet Protocol Network transcode theencoded speech into a lower quality format. The buffering andtransmission characteristics of this connection are adjusted to ensurethat no data packets are lost or assembled out of order.

The VoIP quality speech process can be automatically or manuallyactivated in response to a subscriber accessing a speech qualitysensitive resource or in response to an activation of the feature by thesubscriber, or when it is determined that the originating subscriberterminal device requires the transmission of high quality speechsignals. In order to address the case of lost or damaged packets, theVoIP quality speech system activates a transmit buffer in the portcircuit of the originating system to store the encoded data receivedfrom the transmitting device as it is being output to the network. Thetransmit buffer can then retransmit missing or damaged packets.

The size of the transmit buffer can be controlled by the signalsreceived from the destination system and/or by the network. For example,a speech recognition engine can generate a confidence level, whichmeasure can be returned to the transmitting device to enable the portcircuit that serves the transmitting device to dynamically allocatetransmit buffer resources. In addition, the packets are time stamped andthe difference between the present time and the time stamp on a receivedpacket is an indication of the network delay, which indication can beused by the destination system or the network to control the size of thetransmit buffer. The destination system can monitor receipt of packetsin the jitter buffer to thereby determine in advance of the need for themissing packet to pause the processing of received packets, such as at arational break in the transmission stream—between words, during pauses,between characters in a TTY transmission, etc. When the need for thehigh quality speech signals is satisfied and the continuingcommunications on the call connection can be satisfied by the speedoptimized processing of the speech signals, the VoIP quality speechfeature can be disabled and the quality speech processing resourcesreleased, since the underlying network is by default speed optimized.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates in block diagram form the overall architecture of thepresent VoIP quality speech system and a typical environment, includinga Voice over Internet Protocol Network, in which it is operational; and

FIG. 2 illustrates in flow diagram form the operation of the presentVoIP quality speech system in a typical call scenario.

DETAILED DESCRIPTION OF THE INVENTION

FIG. 1 illustrates in block diagram form the overall architecture of thepresent VoIP quality speech system 142 and a typical environment,including a Voice over Internet Protocol (VoIP) Network 102, in which itis operational. The Internet is an example of an Internet ProtocolNetwork 102 consisting of many computers located all over the world,which are connected through many telephone systems and other means. TheInternet uses a network protocol termed Transmission ControlProtocol/Internet Protocol (TCP/IP). The computers connected to theInternet use TCP/IP to exchange data with other computers that areconnected to the Internet. TCP is a packet switched network protocolthat breaks down the message to be sent into smaller portions of datatermed packets. Each data packet is provided with the data address ofboth the sender and receiver of the message as well as a sequence numberthat identifies where in the sequence of packets this packet belongs.

In a typical telecommunications network environment, the TCP/IP protocolis used to transfer the data from the originating device to thedestination system over the Internet by transmitting each packet for aport circuit 131-133 that serves the originating device through a seriesof routers 102A-D to a port circuit 135-135 that serves the destinationsystem, where the routers 102A-D dynamically interconnect theoriginating device to the destination system. Each router, such asrouter 102A, that receives a data packet examines the destinationaddress and forwards the packet to the next router, such as router 102D,in the Internet Protocol Network 102 to advance the transmission of thedata packet to the destination system. This process continues until thedata packet reaches the destination system.

The individual data packets are reassembled at the destination systemusing TCP to reconstruct the original message. The data packets canarrive out of order at the destination system, since each data packet isdynamically routed by the Internet Protocol Network 102 and likelytraverse different paths, each with its own delay characteristics. Thus,the unique label assigned to each data packet is necessary to ensurethat the data packets which comprise the components of the originalmessage are assembled in the proper order. If the destination systemdetermines that an individual data packet contains an error or a datapacket is missing from the message, it can request the originatingdevice to retransmit the data packet that was identified as eithererroneous or missing.

The expectations with the use of the TCP-IP protocol is that there is anexisting data file resident at the originating device, so there is noneed for a buffer at the originating device, since the entirety of thefile is available at all times during the transmission to retrieve datafor retransmission. Thus, the TCP/IP process ensures accurate, althoughnot real-time, transmission of data from an originating device to adestination system, since the received data must be buffered until thecompleteness of the received data is assured.

In the system illustrated in FIG. 1, a plurality of subscriber terminaldevices 111-113 are served by a switching system, such as a PrivateBranch eXchange (PBX) 101 (or other local switching office) whichinterconnects the subscriber terminal devices 111-113 with a pluralityof trunks 131-133 located in Internet Protocol Network 102 via aswitching network 163 pursuant to the operational signals generated bycontroller 161. The PBX 101 is equipped with a plurality of line portcircuits 121-123, each of which serves a corresponding one of theplurality of subscriber terminal devices 111-113 and a plurality oftrunk port circuits 151-153, each of which serves to connect PBX 101 toa corresponding one of the trunks 131-133 located in Internet ProtocolNetwork 102. Each line port circuit 121-123 is also equipped with atransmission buffer 141-143 to store a predetermined and optionallycontrollable amount of data packets. The VoIP quality speech system 162and the transmit buffers 141-143 are shown as being resident in the PBX101, although their location can be elsewhere in the Internet ProtocolNetwork 102 or in the interconnected devices shown in FIG. 1.

The Internet Protocol Network 102 includes port circuits 135-137, eachof which serves one of a plurality of destination systems 105-107, whichare shown as speech-based systems that use the Internet Protocol Network102 for delivering speech signals from a source subscriber terminaldevice. Examples of destination systems 105-107 include, but are notlimited to: a speech recognition system 107 (which includes a speechrecognition engine 109) that serves a processor 108, voice mail system105, TTY communications (Telecommunication Device for the Deaf device106, and the like.

In each of these cases, the destination system or its associated portcircuit 135-137 includes a jitter buffer, such as jitter buffer 110,that functions to smooth out variance in the data transmission speed bystoring a small amount of the data as it is received before it is neededby the destination system to account for minor variations in the datatransmission speed. The jitter buffer also enables the destinationsystem to assemble a string of packets that are received out of order.However, where the delays in the transmitted data become large, thejitter buffer cannot compensate for these delays and the delays arenoticeable. If the destination system is a unidirectional speech qualitysensitive device, the use of a large jitter buffer solves the delay partof the problem, but cannot address the instance of missing packets.

The packets that are transmitted through an Internet Protocol Network102 can be lost or damaged in the routing through the network. Since thepacket-based Internet Protocol Network 102 dynamically routes eachpacket, there is no guarantee that the packet reaches the destinationsystem in the proper order or even reaches the destination system. Sincethe path through the Internet Protocol Network 102 varies, the onlypoint at which the packetized data can be stored with assurance is inthe originating port circuit or the originating device. To accuratelyreconstitute the missing packet, the originally encoded packets must bebuffered so that retransmission of the missing packet can occur. Thiscan be done, for example, in the originating PBX 101 where the portcircuit 121-123 serving the subscriber terminal equipment 111-113 can beequipped with a transmit buffer 141-143 (alternatively, the portcircuits 131-133 can be equipped with a transmit buffer). The transmitbuffers 141-143 store a predetermined quantity of packets where thecontents of the transmit buffer 141-143 can automatically be aged, suchthat a predetermined quantity of the encoded speech packets are stored,with each successively received packet causing the oldest stored packetto be discarded. In addition, the retransmission scheme can beoriginated only after a predetermined number or frequency of errors isdetected.

Processing of a Call Origination

FIG. 2 illustrates in flow diagram form the operation of the presentVoIP quality speech system 142 and its associated transmit buffers141-143 and jitter buffers 110. The subscriber (calling party) at atypical digital subscriber terminal device, such as 111, initiates atelephone call at step 201 in a standard manner to a destination device,such as the processor 108. In response to the call initiation,controller 141 activates switching network 143 in PBX 101 tointerconnect the port circuit 121 that serves the subscriber terminaldevice 111 with an available trunk circuit 152 at step 202. Thiscommunication connection is extended at step 203 through IP Network 102to the destination, such as speech recognition system 103 that is alsoserved by IP Network 102. The speech recognition system 103 includes aspeech recognition engine 109 that analyses received speech signals todetermine whether the calling party at subscriber terminal device 111 isauthorized to access the protected processor 108 that is served by orpart of speech recognition system 107. Voice quality is especiallycritical for proper operation of the speech recognition engine 109 thatis located in the speech recognition system 107, since the precisedetermination of speech characteristics of the subscriber is essentialto the speech recognition function.

The VoIP quality speech process 142 is activated in response to a oneof: a subscriber accessing a speech quality sensitive resource, such asspeech recognition system 107, or in response to an activation of thefeature by the subscriber, or when it is determined that the originatingsubscriber terminal device requires the transmission of high qualityspeech signals. For the purpose of this description, assume that VoIPquality speech process 142 is automatically activated in response to thecalling party accessing speech recognition system 107.

The present system uses a transmission buffer 141 in the originatingdevice port circuit to store a predetermined and optionally controllableamount of data packets. This system is backwards compatible withexisting telephones and the presence of loss in the network. Sinceexisting UDP devices have no transmit buffer, the transmit buffer 141 islocated in the port circuit 121 that serves the originating device 111.This system encodes the data received from the originating device 121,stores it in the transmit buffer 141 as it is being output to thenetwork 102. The size of the transmit buffer 141, in systems wherebuffer space is shared among a plurality of port circuits 121-123, canbe controlled by the signals received from the destination device and/orby the network 102. For example, a speech recognition engine 109 cangenerate a confidence level, which measure can be returned to theoriginating device 111 to enable the port circuit 121 to allocate bufferresources for the port circuit 121 that serves the originating device.The buffer size can be dynamically adjusted and such a system isbackwards compatible with existing systems.

The data output generated by the subscriber terminal device 111 isstored at step 204 in transmit buffer 141 that is part of or associatedwith the port circuit 121 that serves subscriber terminal device 111.The VoIP quality speech system 142 also signals the Internet ProtocolNetwork 102 at step 205 to transmit the packets of encoded speech usinga non-retransmission protocol by using, as one option, the User DatagramProtocol (UDP) implemented on an Internet Protocol Network 102 tominimize the transmission delays. Thus, the Internet Protocol Network102 processes the received coded signals from the port circuit 121,which coded signals are also stored in transmit buffer circuit 141,without modifying the content of these coded signals or being concernedwith the retransmission of lost or damaged packets.

The coded signals are transmitted at step 206 to the destination device,speech recognition system 107, since the receipt and processing of thecalling party's speech input need not be done in real time, as long asthe quality of the coded speech signals is not degraded by thetransmission process.

Activation of the Packet Retransmission Process

When the number and frequency of the transmission errors exceedpredetermined limit(s) as determined at step 207 by the speechrecognition system 107 (including a threshold where no loss or damage ofpackets is acceptable), the speech recognition system 107 determinesthat the signal quality is unacceptable. Speech recognition system 107accomplishes this in well-known fashion by sequencing the incomingpackets appropriately within it's jitter buffer 110 and monitoring thedata in the jitter buffer 110, looking for gaps in the sequence ofpackets (as indicated by their headers) that indicate the absence ofpackets. When it determines that a packet is missing, the destinationsystem (speech recognition system 107) requests that the originatingsystem retransmit the packet at step 208.

At this juncture, the VoIP quality speech system 142, in response toreceipt of a signal indicating excessive errors, can switch to a pseudoTCP/IP transmission mode with regard to the retransmission of missingpackets, using the buffer circuit 141 of port circuit 121 at step 209 toprovide the speech recognition system 107 with the missing packet(s),each identified by the appropriate header information to enable speechrecognition system 107 to reassemble the stream of information. TheInternet Protocol Network 102 does not change its mode of operation andcontinues to transmit the packets of encoded speech using anon-retransmission protocol, such as UDP, to minimize the transmissiondelays and to not transcode the encoded speech into a lower qualityformat. The retransmission of packets in this mode is effected by theVoIP quality speech system 142, operating in concert with the speechrecognition system 107, in a manner that is transparent to the InternetProtocol Network 102.

Buffer Management Processes

The size of the jitter buffer 110 that is located in speech recognitionsystem 107 is typically selected to avoid interruptions in processing ofthe received packets. This determination of buffer size would typicallyinclude a determination of the amount of time that it takes to identifythe loss of a packet and the time that elapses before a requestedmissing packet can be retransmitted by the VoIP quality speech system142 and received by the speech recognition system 107. Thus, the speechrecognition system 107 may have to adjust how quickly it determines thata packet is missing in the incoming data stream and/or adjust the sizeof the jitter buffer 110 (which would have the effect of altering thelatency).

For example, the speech recognition system 107 determines whether packetloss levels are acceptable, and the extent to which it makes sense totrade latency for reduced packet loss. For example, with TDD/TTYtransmissions, and with a packet size of 20 ms, it is known that theFCC-allowable character error rate is exceeded when packet loss exceeds0.12%. The destination system could therefore adjust the buffer sizeautomatically, such that packet loss would not exceed 0.12%. In a speechrecognition or speaker identification application, recognition accuracyand associated “confidence levels” could be the basis for the jitterbuffer size adjustments.

The size of the transmit buffer 141, located in the port circuit 121associated with the originating device 111, can be controlled by thesignals received from the destination device and/or by the network. Forexample, a speech recognition engine 109 can generate a confidencelevel, which measure can be returned at step 210 to the originatingdevice 111 to enable the port circuit 121 that serves the originatingdevice 111 to dynamically allocate transmit buffer resources. Inaddition, the packets that are transmitted through network 102 are timestamped and the difference between the present time and the time stampon a received packet is an indication of the network transmission delay.Thus, the network 102 or the destination device 107 can use thisindication to control the size of the transmit buffer 141. Thedestination device 107 can monitor receipt of packets in the jitterbuffer 110 to thereby determine, in advance of the need for the missingpacket, to pause the processing of received packets, such as at arational break in the transmission stream—between words, during pauses,between characters in a TTY transmission, etc. When the need for thehigh quality speech signals is satisfied and the continuingcommunications on the call connection can be satisfied by the speedoptimized processing of the speech signals, the VoIP quality speechfeature can be disabled and the quality speech processing resourcesreleased, since the underlying network is by default speed optimized.

Once the need for quality speech transmission has ended, the VoIPquality speech system 142 can return the operation of the systemillustrated in FIG. 1 to the default mode of operation. In particular,at step 211, the speech recognition system 103 can transmit anindication that it has concluded the processing of the speech input. Inresponse thereto, the VoIP quality speech system 142 at step 212terminates the retransmission of lost or damaged packets.

Summary

The VoIP quality speech system is selectively activated when there is aneed for quality speech transmission. The VoIP quality speech system canswitch to retransmission of missing packets, using the buffer circuit ofthe originating port circuit to provide the missing packets. Theretransmission of packets in this mode is effected by the VoIP qualityspeech system, operating in concert with the destination, in a mannerthat is transparent to the Internet Protocol Network.

1. A system for transmitting quality speech signals in a communication connection from an originating device to a destination device over an IP-based network, comprising: a port circuit for transmitting data packets, containing encoded speech signals received from an associated originating device, to said destination device via said IP-based network; transmit buffer means, connected to said port circuit associated with said originating device, for storing a plurality of said data packets received from said associated originating device; network activation means for activating said IP-based network to operate using a packet transmission protocol that fails to retransmit lost or damaged packets; and packet retransmission means, operable independent of said packet transmission protocol, for activating said port circuit to retrieve a lost or damaged packet from said transmit buffer means for retransmission to said destination device.
 2. The system for transmitting quality speech signals of claim 1 wherein said packet retransmission means comprises: packet error detection means, connected to said destination device, for generating an indication that identifies a missing packet; and means for transmitting a signal to said port circuit associated with said originating device requesting retransmission of said identified missing packet.
 3. The system for transmitting quality speech signals of claim 1 further comprising: transmit buffer control means for transmitting a signal to said port circuit associated with said originating device to regulate the size of said transmit buffer means.
 4. The system for transmitting quality speech signals of claim 1 further comprising: jitter buffer management means for regulating a size of a jitter buffer associated with said destination device as a function of at least one of: network transmission delay, speed of processing received packets, time required to identify the absence of a packet in a sequence of received packets, time required to receive a retransmitted packet.
 5. The system for transmitting quality speech signals of claim 1 further comprising: application detection means for determining that said communication connection serves a speech-based application that requires high quality speech signals.
 6. The system for transmitting quality speech signals of claim 5 further comprising: network control means, responsive to said application detection means, for activating said IP-based transmission medium to transmit said high quality digital encoded speech signals without transcoding.
 7. The system for transmitting quality speech signals of claim 5 further comprising: process disabling means, responsive to the conclusion of operation of said speech-based application, for disabling operation of said packet retransmission means.
 8. The system for transmitting quality speech signals of claim 5 wherein said application detection means comprises: destination device identification means for determining the presence of a destination device on said communication connection that requires high quality speech signals.
 9. The system for transmitting quality speech signals of claim 5 wherein said application detection means comprises: registration process detection means for determining the presence of a subscriber identification process at said destination device.
 10. The system for transmitting quality speech signals of claim 9 further comprising: process disabling means, responsive to the conclusion of operation of said subscriber identification process, for disabling operation of said packet retransmission means.
 11. A method for transmitting quality speech signals in a communication connection from an originating device to a destination device over an IP-based transmission medium, comprising: transmitting data packets, containing encoded speech signals received from said originating device, from a port circuit serving said originating device to said destination device via said IP-based network; storing, in a transmit buffer connected to said port circuit, a plurality of said data packets received from said associated originating device; activating said IP-based network to operate using a packet transmission protocol that fails to retransmit lost or damaged packets; and activating, independent of said packet transmission protocol, said port circuit to retrieve a lost or damaged packet from said transmit buffer for retransmission to said destination device.
 12. The method for transmitting quality speech signals of claim 11 wherein said step of activating said port circuit comprises: generating an indication that identifies a missing packet; and transmitting a signal to said port circuit associated with said originating device requesting retransmission of said identified missing packet.
 13. The method for transmitting quality speech signals of claim 11 further comprising: transmitting a signal to said port circuit associated with said originating device to regulate the size of said transmit buffer.
 14. The method for transmitting quality speech signals of claim 11 further comprising: regulating a size of a jitter buffer associated with said destination device as a function of at least one of: network transmission delay, speed of processing received packets, time required to identify the absence of a packet in a sequence of received packets, time required to receive a retransmitted packet.
 15. The method for transmitting quality speech signals of claim 11 further comprising: determining that said communication connection serves a speech-based application that requires high quality speech signals.
 16. The method for transmitting quality speech signals of claim 15 further comprising: activating, in response to said step of determining, said IP-based transmission medium to transmit said high quality digital encoded speech signals without transcoding.
 17. The method for transmitting quality speech signals of claim 16 further comprising: disabling, in response to the conclusion of operation of said speech-based application, operation of said step of activating said port circuit to retransmit lost or damaged packets.
 18. The method for transmitting quality speech signals of claim 16 wherein said step of determining comprises: determining the presence of a destination device on said communication connection that requires high quality speech signals.
 19. The method for transmitting quality speech signals of claim 16 wherein said step of determining comprises: determining the presence of a subscriber identification process at said destination device.
 20. The method for transmitting quality speech signals of claim 19 further comprising: disabling, in response to the conclusion of operation of said subscriber identification process, operation of said step of activating said port circuit to retransmit lost or damaged packets. 21-29. (canceled)
 30. A method for transmitting data signals in a communication connection from an originating device to a destination device over an IP-based network, comprising: transmitting data packets from a first communication device to a second communication device via said IP-based network using a first transmission protocol that does not retransmit transmitted packets that are at least one of lost and damaged; determining that network performance of said IP-based network is insufficient to transmit quality data signals using the first transmission protocol; and changing from transmitting data packets using a first transmission protocol to transmitting data packets using a second transmission protocol that provides for retransmission of transmitted packets that are at least one of lost and damaged.
 31. The method for transmitting data signals of claim 30 wherein said step of switching comprises: generating an indication that identifies a missing packet; and transmitting a signal to said first communication device requesting retransmission of said identified packet.
 32. The method for transmitting data signals of claim 31 further comprising: transmitting a signal to said first communication device to regulate size of said transmit buffer.
 33. The method for transmitting data signals of claim 31 further comprising: regulating a size of a jitter buffer associated with said second communication device as a function of at least one of: network transmission delay, speed of processing received packets, time required to identify absence of a packet in a sequence of received packets, and time required to receive a retransmitted packet.
 34. The method for transmitting data signals of claim 31 further comprising: determining that said communication connection serves a speech-based application that requires high quality audio signals.
 35. The method for transmitting data signals of claim 34 further comprising: activating, in response to said step of determining, said IP-based network to transmit said high quality digital encoded speech signals without transcoding.
 36. The method for transmitting data signals of claim 35 further comprising: disabling, in response to the conclusion of operation of said speech-based application, operation of said step of activating said first communication device to retransmit transmitted packets that are lost or damaged.
 37. The method for transmitting data signals of claim 35 wherein said step of determining comprises: determining presence of a second communication device on said communication connection that requires high quality audio signals.
 38. The method for transmitting data signals of claim 35 wherein said step of determining comprises: determining presence of a subscriber identification process at said second communication device.
 39. The method for transmitting data signals of claim 38 further comprising: disabling, in response to conclusion of operation of said subscriber identification process, operation of said step of activating said port circuit to retransmit transmitted packets that are lost or damaged. 40-44. (canceled) 